Telephone carriers are deploying various packet-based voice technologies such as Real-time Transport Protocol/Internet Protocol (RTP/IP) and Asynchronous Transfer Mode Adaptation Layer 2 (ATM/AAL2). These technologies do not interwork seamlessly. Currently deployed call agents, software systems that establish the connections across packet-based voice network, do not have the capability to co-ordinate the conversion of different types of packet-based data. Two different types of packet-based voice technologies, for example VoIP and VoATM can be made to interwork with each other with a public switched telephone network (PSTN) between them. However, the PSTN middleman necessitates costly and inefficient conversion into outdated time division multiplexing (TDM) format and thence to another packet-based format. This lack of interworking is at the bearer and control levels. As a result, there are various, pioneering packet “islands” that use the outdated, PSTN as the glue, thereby annulling the advantages of packet-based voice technology over large geographical areas. Using the PSTN also incurs signal degradation because the PSTN uses only non-compressed voice signals. Packet networks may use compressed signals that need to be converted into non-compressed format and then converted back into compressed format.
FIG. 1 is a block diagram depicting a typical conversion from an IP network to an ATM network. In the telecommunications network 100 shown in FIG. 1, telephonic data is received at voice over IP (VoIP) edge gateway 102. This data may be received from individual telephones, a private telephone network such as a private branch exchange (PBX), a data modem, or a fax machine, among others. Edge gateway 102 is a combination of software and hardware that bridges the gap between the telephone network and the IP network. Edge gateway 102 may be integrated into the telephone or PBX. The telephonic data is then routed over IP network 104 to trunk gateway 106. Establishment of the connection between the VoIP edge gateway 102 and trunk gateway 106 is controlled by one, or more, call agents 108. The call agent 108 establishes the IP session between the VoIP edge gateway 102 and the trunk gateway 106, and coordinates the conversion of data from IP format to TDM format. The data is transmitted over TDM trunk lines 109 to a network of PSTN switches 110. The TDM trunk lines may be, for example, T1 lines. The data is now transmitted over TDM trunk lines 111 to trunk gateway 112. The connection between the trunk gateway 112 and the voice over ATM (VoATM) edge gateway 116 is controlled by one, or more, call agents 118. Further, call agents 108 and 118 can communicate with each other and with the PSTN switches through a Signaling System 7 (SS7) control network. The call agent 118 initiates the establishment of an ATM connection, and coordinates the conversion of data from TDM format to ATM format. The data is routed through ATM network 114 to VoATM edge gateway 116. From VoATM gateway 116 the telephonic data is transmitted to its destination telephone or PBX, for example.
The routing of packet-based voice data through a PSTN defeats one of the advantages of packet-based voice transmission, which is that the voice data can be compressed, thereby reducing bandwidth and cost. No such voice compression is possible in a PSTN; the telephonic data must be decompressed upon entering the PSTN and, recompressed upon exiting the PSTN. By routing VoIP data through a PSTN to an ATM network, this major advantage of packet-based voice technology is negated.